LiveKit docs › Telephony › SIP API --- # SIP APIs > Use LiveKit's built-in SIP APIs to manage your SIP-based apps. ## Overview LiveKit has built-in APIs that let you manage SIP trunks, dispatch rules, and SIP participants. These APIs are available with LiveKit server SDKs and CLI: - [Go SIP client](https://pkg.go.dev/github.com/livekit/server-sdk-go/v2#SIPClient) - [JS SIP client](https://docs.livekit.io/reference/server-sdk-js/classes/SipClient.html.md) - [Ruby SIP client](https://github.com/livekit/server-sdk-ruby/blob/main/lib/livekit/sip_service_client.rb) - [Python SIP client](https://docs.livekit.io/reference/python/livekit/api/sip_service.html.md) - [Java SIP client](https://github.com/livekit/server-sdk-kotlin/blob/main/src/main/kotlin/io/livekit/server/SipServiceClient.kt) - [CLI](https://github.com/livekit/livekit-cli/blob/main/cmd/lk/sip.go) > ❗ **Important** > > Requests to the SIP API require the SIP `admin` permission unless otherwise noted. To create a token with the appropriate grant, see [SIP grant](https://docs.livekit.io/frontends/authentication/tokens.md#sip-grant). For phone number management APIs, see [Phone Number APIs](https://docs.livekit.io/reference/telephony/phone-numbers-api.md). To learn more about additional APIs, see [Server APIs](https://docs.livekit.io/reference/server/server-apis.md). ### Using endpoints The SIP API is accessible via `/twirp/livekit.SIP/`. For example, if you're using LiveKit Cloud the following URL is for the [CreateSIPInboundTrunk](#createsipinboundtrunk) API endpoint: ```shell https://%{projectDomain}%/twirp/livekit.SIP/CreateSIPInboundTrunk ``` #### Authorization header All endpoints require a signed access token. This token should be set via HTTP header: ``` Authorization: Bearer ``` LiveKit server SDKs automatically include the above header. #### Post body Twirp expects an HTTP POST request. The body of the request must be a JSON object (`application/json`) containing parameters specific to that request. Use an empty `{}` body for requests that don't require parameters. #### Examples For example, create an inbound trunk using [CreateSIPInboundTrunk](#createsipinboundtrunk): ```shell curl -X POST https://%{projectDomain}%/twirp/livekit.SIP/CreateSIPInboundTrunk \ -H "Authorization: Bearer " \ -H 'Content-Type: application/json' \ -d '{ "name": "My trunk", "numbers": ["+15105550100"] }' ``` Use the [ListSIPInboundTrunk](#listsipinboundtrunk) API endpoint to list inbound trunks: ```shell curl -X POST https://%{projectDomain}%/twirp/livekit.SIP/ListSIPInboundTrunk \ -H "Authorization: Bearer " \ -H 'Content-Type: application/json' \ -d '{}' ``` ## SIPService APIs The SIPService APIs allow you to manage trunks, dispatch rules, and SIP participants. > 💡 **Tip** > > All RPC definitions and options can be found [here](https://github.com/livekit/protocol/blob/main/protobufs/livekit_sip.proto). ### CreateSIPInboundTrunk Create an inbound trunk with the specified settings. Returns [SIPInboundTrunkInfo](#sipinboundtrunkinfo). | Parameter | Type | Required | Description | | name | string | yes | Name of the trunk. | | metadata | string | | Initial metadata to assign to the trunk. This metadata is added to every SIP participant that uses the trunk. | | numbers | array | yes | Array of provider phone numbers associated with the trunk. | | allowed_addresses | array | | List of IP addresses that are allowed to use the trunk. Each item in the list can be an individual IP address or a Classless Inter-Domain Routing notation representing a CIDR block. | | allowed_numbers | array | | List of phone numbers that are allowed to use the trunk. If this [list is empty](https://docs.livekit.io/telephony/accepting-calls/inbound-trunk.md#accepting-calls-to-any-phone-number), trunk access must be limited by using one of the following options: - Set `auth_username` and `auth_password` _or_ - Set `allowed_addresses` | | auth_username | string | | If configured, the username for authorized use of the provider's SIP trunk. | | auth_password | string | | If configured, the password for authorized use of the provider's SIP trunk. | | headers | map | | SIP X-* headers for INVITE request. These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint. | | headers_to_attributes | map | | Key-value mapping of SIP X-* header names to participant attribute names. | | attributes_to_headers | map | | Map SIP headers from INVITE request to `sip.h.*` participant attributes. If the names of the required headers is known, use `headers_to_attributes` instead. | | include_headers | [SIPHeaderOptions](#sipheaderoptions) | | Specify how SIP headers should be mapped to attributes. | | ringing_timeout | [google.protobuf.Duration](https://github.com/protocolbuffers/protobuf/blob/main/src/google/protobuf/duration.proto) | | Maximum time for the call to ring. | | max_call_duration | google.protobuf.Duration | | Maximum call duration. | | krisp_enabled | bool | | True to enable [Krisp noise cancellation](https://docs.livekit.io/transport/media/noise-cancellation.md#telephony) for the caller. | | media_encryption | [SIPMediaEncryption](#sipmediaencryption) | | Whether or not to encrypt media. | ### CreateSIPOutboundTrunk Create an outbound trunk with the specified settings. Returns [SIPOutboundTrunkInfo](#sipoutboundtrunkinfo). | Parameter | Type | Required | Description | | name | string | yes | Name of the trunk. | | metadata | string | | User-defined metadata for the trunk. This metadata is added to every SIP participant that uses the trunk. | | address | string | yes | Hostname or IP the SIP INVITE is sent to. This is _not_ a SIP URI and shouldn't contain the `sip:` protocol. | | destination_country | string | yes | Two-letter [country code](https://en.wikipedia.org/wiki/ISO_3166-1_alpha-2) for the country the call terminates in. LiveKit uses the country code to route calls. To learn more, see [Outbound calls](https://docs.livekit.io/telephony/features/region-pinning.md#outbound-calls). | | numbers | array | yes | List of provider phone numbers associated with the trunk that can be used as a caller id. | | transport | [SIPTransport](#siptransport) | | Protocol to use for SIP transport: auto, TCP, or UDP. | | auth_username | string | | If configured, the username for authorized use of the provider's SIP trunk. | | auth_password | string | | If configured, the password for authorized use of the provider's SIP trunk. | | headers | map | | SIP X-* headers for INVITE request. These headers are sent as-is and may help identify this call as coming from LiveKit for the other SIP endpoint. | | headers_to_attributes | map | | Key-value mapping of SIP X-* header names to participant attribute names. | | media_encryption | [SIPMediaEncryption](#sipmediaencryption) | | Whether or not to encrypt media. | ### CreateSIPDispatchRule Create dispatch rule. Returns [SIPDispatchRuleInfo](#sipdispatchruleinfo). | Parameter | Type | Required | Description | | dispatch_rule | [SIPDispatchRuleInfo](#sipdispatchruleinfo) | yes | Dispatch rule to create. | | trunk_ids | array | | List of associated trunk IDs. If empty, all trunks match this dispatch rule. | | hide_phone_number | bool | | If true, use a random value for participant identity and phone number omitted from attributes. By default, the participant identity is created using the phone number (if the participant identity isn't explicitly set). | | inbound_numbers | array | | If this list is populated, the dispatch rule only accepts calls made _from_ numbers in the list. If a caller's phone number is not in the list, the call is rejected. | | name | string | yes | Human-readable name for the dispatch rule. | | metadata | string | | Optional metadata for the dispatch rule. If defined, participants created by the rule inherit this metadata. | | attributes | map | | Key-value mapping of user-defined attributes. Participants created by this rule inherit these attributes. | | room_preset | string | | Only for LiveKit Cloud: Config preset to use. | | room_config | [RoomConfiguration](https://docs.livekit.io/reference/server/server-apis.md#roomconfiguration) | | Room configuration to use if the participant initiates the room. | ### CreateSIPParticipant > ℹ️ **Note** > > Requires SIP `call` grant on authorization token. Create a SIP participant to make outgoing calls. Returns [SIPParticipantInfo](#sipparticipantinfo) | Parameter | Type | Required | Description | | sip_trunk_id | string | | ID for SIP trunk used to dial user. Required if `trunk` is not set. | | trunk | [SIPOutboundConfig](#sipoutboundconfig) | | Inline outbound trunk configuration. Required if `sip_trunk_id` is not set. When using this parameter, `sip_number` must also be set. | | sip_call_to | string | yes | Phone number to call. | | sip_number | string | | SIP number to call from. Required if using an [inline trunk configuration](https://docs.livekit.io/telephony/making-calls/outbound-calls.md#inline-trunk). If `sip_trunk_id` is set, this defaults to the number associated with the stored trunk. | | room_name | string | yes | Name of the room to connect the participant to. | | participant_identity | string | | Identity of the participant. | | participant_name | string | | Name of the participant. | | participant_metadata | string | | User-defined metadata that is attached to created participant. | | participant_attributes | map | | Key-value mapping of user-defined attributes to attach to created participant. | | dtmf | string | | DTMF digits (extension codes) to use when making the call. Use character `w` to add a 0.5 second delay. | | play_dialtone | bool | | Optionally play dial tone in the room as an audible indicator for existing participants. | | hide_phone_number | bool | | If true, use a random value for participant identity and phone number omitted from attributes. By default, the participant identity is created using the phone number (if the participant identity isn't explicitly set). | | headers | map | | SIP X-* headers for INVITE request. These headers are sent as-is and may help identify this call as coming from LiveKit. | | include_headers | [SIPHeaderOptions](#sipheaderoptions) | | Specify how SIP headers should be mapped to attributes. | | ringing_timeout | google.protobuf.Duration | | Maximum time for the callee to answer the call. The upper limit is 80 seconds. | | max_call_duration | google.protobuf.Duration | | Maximum call duration. | | krisp_enabled | bool | | True to enable [Krisp noise cancellation](https://docs.livekit.io/transport/media/noise-cancellation.md#telephony) for the callee. | | media_encryption | [SIPMediaEncryption](#sipmediaencryption) | | Whether or not to encrypt media. | | wait_until_answered | bool | | If true, return after the call is answered — including if it goes to voicemail. | | display_name | string | | Optional display name for the caller's number. If empty, results in a CNAM lookup by destination carrier. If omitted, defaults to caller's number. For more information, see [Custom caller ID](https://docs.livekit.io/telephony/making-calls/outbound-calls.md#custom-caller-id). | ### DeleteSIPDispatchRule Delete a dispatch rule. Returns [SIPDispatchRuleInfo](#sipdispatchruleinfo). | Parameter | Type | Required | Description | | sip_dispatch_rule_id | string | | ID of dispatch rule. | ### DeleteSIPTrunk Delete a trunk. Returns [SIPTrunkInfo](#siptrunkinfo). | Parameter | Type | Required | Description | | sip_trunk_id | string | yes | ID of trunk. | ### GetSIPInboundTrunk Get inbound trunk. Returns [GetSIPInboundTrunkResponse](#getsipinboundtrunkresponse). | Parameter | Type | Required | Description | | sip_trunk_id | string | yes | ID of trunk. | ### GetSIPOutboundTrunk Get outbound trunk. Returns [GetSIPOutboundTrunkResponse](#getsipoutboundtrunkresponse). | Parameter | Type | Required | Description | | sip_trunk_id | string | yes | ID of trunk. | ### ListSIPDispatchRule List dispatch rules. Returns array<[SIPDispatchRuleInfo](#sipdispatchruleinfo)>. ### ListSIPInboundTrunk List inbound trunks. Returns array<[SIPInboundTrunkInfo](#sipinboundtrunkinfo)>. ### ListSIPOutboundTrunk List outbound trunks. Returns array<[SIPOutboundTrunkInfo](#sipoutboundtrunkinfo)>. ### TransferSIPParticipant > ℹ️ **Note** > > Requires SIP `call` grant on authorization token. Transfer call to another number or SIP endpoint. Returns [google.protobuf.Empty](https://github.com/protocolbuffers/protobuf/blob/main/src/google/protobuf/empty.proto). | Parameter | Type | Required | Description | | participant_identity | string | yes | Identity of the participant to transfer. | | room_name | string | yes | Name of the room the participant is currently in. | | transfer_to | string | yes | Phone number or SIP endpoint to transfer participant to. This value can either be a valid telephone number or a SIP URI. The following examples are valid values: - `tel:+15105550100` - `sip:+15105550100@sip.telnyx.com` - `sip:+15105550100@my-livekit-demo.pstn.twilio.com` | | play_dialtone | bool | | Optionally play dial tone during the transfer. By default, the room audio is played during the transfer. | ### UpdateSIPDispatchRule Update a dispatch rule. Returns [SIPDispatchRuleInfo](#sipdispatchruleinfo). | Parameter | Type | Required | Description | | req | [UpdateSIPDispatchRuleRequest](#updatesipdispatchrulerequest) | yes | Update or replace request. | ### UpdateSIPInboundTrunk Update an inbound trunk. Returns [SIPInboundTrunkInfo](#sipinboundtrunkinfo). | Parameter | Type | Required | Description | | req | [UpdateSIPInboundTrunkRequest](#updatesipinboundtrunkrequest) | yes | Update or replace request. | ### UpdateSIPOutboundTrunk Update an outbound trunk. Returns [SIPOutboundTrunkInfo](#sipoutboundtrunkinfo). | Parameter | Type | Required | Description | | req | [UpdateSIPOutboundTrunkRequest](#updatesipoutboundtrunkrequest) | yes | Update or replace request. | ## Types The SIP service includes the following types. ### GetSIPInboundTrunkResponse | Field | Type | Description | | trunk | [SIPInboundTrunkInfo](#sipinboundtrunkinfo) | Inbound trunk. | ### GetSIPOutboundTrunkResponse | Field | Type | Description | | trunk | [SIPOutboundTrunkInfo](#sipoutboundtrunkinfo) | Outbound trunk. | ### ListUpdate | Field | Type | Description | | add | array | Optional set of items to add. | | set | array | Optional set of items to set. If this field is present, it replaces the existing list. | | remove | array | Optional set of items to remove. | ### SIPDispatchRule Valid values include: | Name | Type | Value | Description | | dispatch_rule_direct | SIPDispatchRuleDirect | 1 | Dispatches callers into an existing room. You can optionally require a pin before caller enters the room. | | dispatch_rule_individual | SIPDispatchRuleIndividual | 2 | Creates a new room for each caller. | | dispatch_rule_callee | SIPDispatchRuleCallee | 3 | Creates a new room for each callee. | ### SIPDispatchRuleDirect | Field | Type | Description | | room_name | string | Name of the room that callers join. | | pin | string | Optional pin a caller must enter to join the room. | ### SIPDispatchRuleIndividual | Field | Type | Description | | room_prefix | string | Prefix added to the generated room name. When set, the room name is `_`; when empty, the room name is the caller's number. | | pin | string | Optional pin a caller must enter to join the room. | ### SIPDispatchRuleCallee | Field | Type | Description | | room_prefix | string | Prefix added to the generated room name. When set, the room name is `_`; when empty, the room name is the called number. | | pin | string | Optional pin a caller must enter to join the room. | | randomize | bool | If true, append a random suffix to the room name so each caller gets a unique room. Defaults to false, which places callers to the same number in the same room. | ### SIPHeaderOptions Enum. Valid values are as follows: | Name | Value | Description | | SIP_NO_HEADERS | 0 | Don't map any headers except those mapped explicitly. | | SIP_X_HEADERS | 1 | Map all `X-*` headers to `sip.h.*` attributes. | | SIP_ALL_HEADERS | 2 | Map all headers to `sip.h.*` attributes. | ### SIPDispatchRuleInfo | Field | Type | Description | | sip_dispatch_rule_id | string | Dispatch rule ID. | | rule | [SIPDispatchRule](#sipdispatchrule) | Type of dispatch rule. | | trunk_ids | array | List of associated trunk IDs. | | hide_phone_number | bool | If true, hides phone number. | | inbound_numbers | array | If this list is populated, the dispatch rule only accepts calls made _from_ numbers in the list. If a caller's phone number is not in the list, the call is rejected. | | name | string | Human-readable name for the dispatch rule. | | metadata | string | User-defined metadata for the dispatch rule. Participants created by this rule inherit this metadata. | | headers | map | Custom SIP X-* headers to include in the 200 OK response. | | attributes | map | Key-value mapping of user-defined attributes. Participants created by this rule inherit these attributes. | | room_preset | string | Only for LiveKit Cloud: Config preset to use. | | room_config | [RoomConfiguration](https://docs.livekit.io/reference/server/server-apis.md#roomconfiguration) | Room configuration object associated with the dispatch rule. | ### SIPDispatchRuleUpdate | Field | Type | Description | | trunk_ids | array | [ListUpdate](#listupdate) | List of trunk IDs to associate with the dispatch rule. | | rule | [SIPDispatchRule](#sipdispatchrule) | Type of dispatch rule. | | name | string | Human-readable name for the dispatch rule. | | metadata | string | User-defined metadata for the dispatch rule. Participants created by this rule inherit this metadata. | | attributes | map | Key-value mapping of user-defined attributes. Participants created by this rule inherit these attributes. | | media_encryption | [SIPMediaEncryption](#sipmediaencryption) | Whether or not to encrypt media. | ### SIPInboundTrunkInfo | Field | Type | Description | | sip_trunk_id | string | Trunk ID | | name | string | Human-readable name for the trunk. | | numbers | array | [ListUpdate](#listupdate) | Phone numbers associated with the trunk. The trunk only accepts calls made to the phone numbers in the list. | | allowed_addresses | array | IP addresses or CIDR blocks that are allowed to use the trunk. If this list is populated, the trunk only accepts traffic from the IP addresses in the list. | | allowed_numbers | array | Phone numbers that are allowed to dial in. If this list is populated, the trunk only accepts calls from the numbers in the list. | | auth_username | string | Username used to authenticate inbound SIP invites. | | auth_password | string | Password used to authenticate inbound SIP invites. | | headers | map | Custom SIP X-* headers to include in the 200 OK response. | | headers_to_attributes | map | Custom SIP X-* headers that map to SIP participant attributes. | | ringing_timeout | [google.protobuf.Duration](https://github.com/protocolbuffers/protobuf/blob/main/src/google/protobuf/duration.proto) | Maximum time for the caller to wait for track subscription (that is, for the call to be picked up). | | max_call_duration | google.protobuf.Duration | Maximum call duration. | | krisp_enabled | Boolean | True if Krisp noise cancellation is enabled for the call. | ### SIPInboundTrunkUpdate | Field | Type | Description | | numbers | array | [ListUpdate](#listupdate) | List of phone numbers associated with the trunk. | | allowed_addresses | array | [ListUpdate](#listupdate) | List of IP addresses or CIDR blocks that are allowed to use the trunk. | | allowed_numbers | array | [ListUpdate](#listupdate) | List of phone numbers that are allowed to use the trunk. | | auth_username | string | Username used to authenticate inbound SIP invites. | | auth_password | string | Password used to authenticate inbound SIP invites. | | name | string | Human-readable name for the trunk. | | metadata | string | User-defined metadata for the trunk. | | media_encryption | [SIPMediaEncryption](#sipmediaencryption) | Whether or not to encrypt media. | ### SIPOutboundConfig Inline outbound trunk configuration for use with [`CreateSIPParticipant`](#createsipparticipant). For more information, see [Inline trunk configuration](https://docs.livekit.io/telephony/making-calls/outbound-calls.md#inline-trunk). | Field | Type | Description | | hostname | string | SIP server address (for example, `.pstn.twilio.com` or `sip.telnyx.com`). | | destination_country | string | Two-letter [country code](https://en.wikipedia.org/wiki/ISO_3166-1_alpha-2) for the country the call terminates in. LiveKit uses the country code to route calls. To learn more, see [Outbound calls](https://docs.livekit.io/telephony/features/region-pinning.md#outbound-calls). | | transport | [SIPTransport](#siptransport) | SIP transport used for the outbound call. | | auth_username | string | Username for authenticated use of the SIP trunk. Can be empty for unauthenticated trunks. | | auth_password | string | Password for authenticated use of the SIP trunk. Can be empty for unauthenticated trunks. | | headers_to_attributes | map | Map of SIP X-* headers from a `200 OK` response to SIP participant attributes. Keys are header names and values are the participant attribute names they map to. | | attributes_to_headers | map | Map of LiveKit attributes to SIP X-* headers when sending BYE or REFER requests. Keys are attribute names and values are the X-* header names they map to. | | from_host | string | Optional custom hostname for the `From` SIP header. When set, outbound calls use this host instead of the default project SIP domain. | ### SIPOutboundTrunkInfo | Field | Type | Description | | sip_trunk_id | string | Trunk ID. | | name | string | Trunk name. | | metadata | string | User-defined metadata for trunk. | | address | string | Hostname or IP address the SIP request message (SIP INVITE) is sent to. | | destination_country | string | Two-letter [country code](https://en.wikipedia.org/wiki/ISO_3166-1_alpha-2) for the country the call terminates in. LiveKit uses the country code to route calls. To learn more, see [Outbound calls](https://docs.livekit.io/telephony/features/region-pinning.md#outbound-calls). | | transport | [SIPTransport](#siptransport) | Protocol to use for SIP transport: auto, TCP, or UDP. | | numbers | array | Phone numbers used to make calls. A random number in the list is selected whenever a call is made. | | auth_username | string | Username used to authenticate with the SIP server. | | auth_password | string | Password used to authenticate with the SIP server. | | headers | map | Custom SIP X-* headers to include in the 200 OK response. | | headers_to_attributes | map | Custom SIP X-* headers that map to SIP participant attributes. | ### SIPOutboundTrunkUpdate | Field | Type | Description | | address | string | Hostname or IP address the SIP request message (SIP INVITE) is sent to. | | transport | [SIPTransport](#siptransport) | Protocol to use for SIP transport: auto, TCP, or UDP. | | destination_country | string | Two-letter [country code](https://en.wikipedia.org/wiki/ISO_3166-1_alpha-2) for the country the call terminates in. LiveKit uses the country code to route calls. To learn more, see [Outbound calls](https://docs.livekit.io/telephony/features/region-pinning.md#outbound-calls). | | numbers | array | [ListUpdate](#listupdate) | Phone numbers used to make calls. A random number in the list is selected whenever a call is made. | | auth_username | string | Username used to authenticate with the SIP server. | | auth_password | string | Password used to authenticate with the SIP server. | | name | string | Human-readable name for the trunk. | | metadata | string | User-defined metadata for the trunk. | | media_encryption | [SIPMediaEncryption](#sipmediaencryption) | Whether or not to encrypt media. | ### SIPParticipantInfo | Field | Type | Description | | participant_id | string | Participant ID. | | participant_identity | string | Participant name. | | room_name | string | Name of the room. | | sip_call_id | string | SIP call ID. | ### SIPMediaEncryption Enum. Valid values are as follows: | Name | Value | Description | | SIP_MEDIA_ENCRYPT_DISABLE | 0 | Don't turn on encryption. | | SIP_MEDIA_ENCRYPT_ALLOW | 1 | Use encryption if available. | | SIP_MEDIA_ENCRYPT_REQUIRE | 2 | Require encryption. | ### SIPTransport Enum. Valid values are as follows: | Name | Value | Description | | SIP_TRANSPORT_AUTO | 0 | Detect automatically. | | SIP_TRANSPORT_UDP | 1 | UDP | | SIP_TRANSPORT_TCP | 2 | TCP | | SIP_TRANSPORT_TLS | 3 | TLS | ### SIPTrunkInfo > ℹ️ **Note** > > This type is deprecated. See [SIPInboundTrunkInfo](#sipinboundtrunkinfo) and [SIPOutboundTrunkInfo](#sipoutboundtrunkinfo). | Field | Type | Description | | sip_trunk_id | string | Trunk ID. | | kind | [TrunkKind](#trunkkind) | Type of trunk. | | inbound_addresses | array | IP addresses or CIDR blocks that are allowed to use the trunk. If this list is populated, the trunk only accepts traffic from the IP addresses in the list. | | outbound_address | string | IP address that the SIP INVITE is sent to. | | outbound_number | string | Phone number used to make outbound calls. | | transport | [SIPTransport](#siptransport) | Protocol to use for SIP transport: auto, TCP, or UDP. | | inbound_numbers | array | List of provider phone numbers this trunk is associated with. Only applies to inbound trunks. If this list is populated, the trunk only accepts calls to numbers in this list. | | inbound_username | string | Username used to authenticate inbound SIP invites. | | inbound_password | string | Password used to authenticate inbound SIP invites. | | outbound_username | string | Username used to authenticate outbound SIP invites. | | outbound_password | string | Password used to authenticate outbound SIP invites. | | name | string | Trunk name. | | metadata | string | Initial metadata to assign to the trunk. This metadata is added to every SIP participant that uses the trunk. | ### TrunkKind Enum. Valid values are as follows: | Name | Value | Description | | TRUNK_LEGACY | 0 | Legacy trunk. | | TRUNK_INBOUND | 1 | [Inbound trunk](https://docs.livekit.io/telephony/accepting-calls/inbound-trunk.md). | | TRUNK_OUTBOUND | 2 | [Outbound trunk](https://docs.livekit.io/telephony/making-calls/outbound-trunk.md). | ### UpdateSIPDispatchRuleRequest | Field | Type | Description | | sip_dispatch_rule_id | string | Dispatch rule ID. | | action | [SIPDispatchRule](#sipdispatchrule) | [SIPDispatchRuleUpdate](#sipdispatchruleupdate) | Dispatch rule for replacement or update. | ### UpdateSIPInboundTrunkRequest | Field | Type | Description | | sip_trunk_id | string | Trunk ID. | | action | [SIPInboundTrunkInfo](#sipinboundtrunkinfo) | [SIPInboundTrunkUpdate](#sipinboundtrunkupdate) | Trunk info for replacement or update. | ### UpdateSIPOutboundTrunkRequest | Field | Type | Description | | sip_trunk_id | string | Trunk ID. | | action | [SIPOutboundTrunkInfo](#sipoutboundtrunkinfo) | [SIPOutboundTrunkUpdate](#sipoutboundtrunkupdate) | Trunk info for replacement or update. | --- This document was rendered at 2026-06-07T11:33:36.391Z. For the latest version of this document, see [https://docs.livekit.io/reference/telephony/sip-api.md](https://docs.livekit.io/reference/telephony/sip-api.md). To explore all LiveKit documentation, see [llms.txt](https://docs.livekit.io/llms.txt).